Telephony is one of the most important inventions in mankind. Since its birth Mar. 10, 1876, installing copper wires to each and everyone that needed communication capabilities has spread the technology worldwide. By coupling copper wires together between caller and called, a connection between these was achieved and they could eventually communicate with each other through their circuit. This kind of technology has become known as circuit-switched telephony. Anyone familiar with classical telephony know that there has been a great evolution within this circuit-switched telephony with, for instance, the AXE platform the Ericsson Corporation developed as their switching solutions. Knowledge about statistical multiplexing of calls within the networks has make it possible to build networks with worldwide coverage to limited costs.
During the last decade, the classical circuit-switched telephony service has met a competitor in the more cost efficient packet-switched telephony built upon the Internet protocol suite Transport Control Protocol/Internet Protocol (TCP)/(IP). This telephony is usually referred to as IP telephony, which currently is being standardized and frequently installed instead of the old circuit-switched telephony.
The packet-switched IP telephony networks are commonly routed using one of the well-known IP routing protocols such as OSPF (Moy J., OSPF Version 2, IETF, RFC2328), IS-IS (Oran D., OSI IS-IS Intra-domain Routing Protocol, IETF, RFC1142) or RIP (Malkin G., RIP Version 2, IETF, RFC2453). These protocols can be classified either as being link-state or distance vector protocols based on the algorithms they use for route computation and distribution of routing information. All routers running a link state protocol within a domain have a complete view of the network, knowing all the networks and routers within the domain. A distance vector router knows only the routers and networks in its immediate surrounding (directly connected).
Most commercial IP telephony systems follow the International Telecommunication Union-Telephony (ITU-T) Recommendation H.323. This recommendation was early adopted by major IP telephony vendors in their systems solutions. In FIG. 1, an overview of the major components in an H.323 system is shown.
These major components are terminals T, gateways G, and gatekeepers Gk. The three first components are referred to as endpoints of the H.323 system since these can initiate or terminate media streams. The gatekeeper is the manager of the H.323 system. The managing domain is referred to as a zone. There is one, and only one, gatekeeper available in each zone.
Terminals
Terminals T are endpoints that provide real-time two-way communications, i.e. it is possible to talk and listen to another H.323 terminal T or another telecommunications system via a gateway. It can also participate in a multipoint conference through the MCU, which will be introduced below. An H.323 terminal T must support the voice service. Besides the voice service, the terminal T can also provide video and data services, but these are optional. To be able to negotiate channel usage and do capability exchange between end-points, the terminal T must also support H.245. Other required components are call setup and signalling via Q.931, registration/admission/status (RAS) for gatekeeper communication, and RTP/RTCP for transportation of real-time services, e.g. voice and video.
Besides these required components, the terminal T could also have MCU capabilities.
Gateway
A gateway G is an interface between an H.323 system and another telecommunication systems, e.g. PSTN. The gateway G is optional and is only required when an endpoint communicates with other terminal types 102, e.g. ISDN, PSTN etc. The gateway G handles both the call control and the call transportation translation between the H.323 system and the non-H.323 system.
Multipoint Control Units
The multipoint control units (MCU) support conferences between three or more endpoints. The MCU comprises a mandatory multipoint controller (MC) and optionally one or more multipoint processors (MP). The MC can be co-located with another end-point, e.g. in a terminal. The MC handles negotiations between terminals during audio and video capability exchange. The MC also determines if any of the related media streams should be distributed with multicast. In case mixing of media streams are required, the MP handles this. As depicted in FIG. 2, Multipoint communication can be made either in a centralised or a decentralised manner. In centralised multipoint conferencing, all communication between endpoints E is made via the MC. In the MC, the media streams are mixed together and distributed to involved endpoints E. In decentralised multipoint conferencing, the MC handles the negotiations while the endpoints E them self distributes the media streams. This distribution can be made with the resource efficient technology multicast. A mesh of unicast media distribution can also be used. Besides the centralised and the decentralised distribution methods, hybrids between these are possible.
Gatekeeper
The gatekeeper Gk is the most important component of an H.323 enabled network. It performs two important call control functions; address translation and bandwidth management. Address translation means that the gatekeeper Gk translates from aliases for terminals and gateways to IP addresses. The bandwidth management implementation is vendor specific. A commonly used method is to specify a threshold for the number of simultaneous calls that can be made within the zone the gatekeeper Gk manages. Other methods might exist but these are in that case vendor specific. Calls can be made directly between endpoints or via the gatekeeper Gk. The latter is referred to as gatekeeper-routed calls.
Even though H.323 primarily was developed for non-guaranteed quality of service networks, the recommendation has been expanded to cover Quality of Service (QoS) issues as well. For instance, QoS Support for H.323 using RSVP is discussed in appendix II of ITU-T Recommendation H.323 version 2, Packet-based multimedia communication systems, Gonova, 1997.
Lack of topology awareness and path sensitive admission control is the most important drawback of current implementations of H.323 gatekeepers. In FIG. 3, a topology for an H.323 enabled network 300 is shown. In the network there are three edge routers ER, one gatekeeper Gk and one gateway G to PSTN. In this network the routers R that connects LAN segments are geographically distributed, e.g. Stockholm, Gothenburg, Malmo, etc. Between these geographically distributed routers R the bandwidth is limited. The gatekeeper Gk manages the whole network which then defines a zone. The gatekeeper Gk and the gateway G are geographically located where the number of users is highest. The gatekeeper GK could be located anywhere, but for practical reasons these are co-located. The logical location for the gateway G is to place it where most of the calls are made. This will result in less routing of calls through the rest of the network, which would be the case if the gateway G were located in a LAN segment far away from the majority of the users.
The gatekeeper Gk can be configured to allow X simultaneous calls on a heuristic basis. If the perceived quality is degraded, the threshold of simultaneous calls can be decreased. This heuristic decision base will cause problems. One user can, with or without malicious intentions, cause low overall utilisation and denial of service to other users. By starting sessions that use a thin bottleneck link, the heutistics will be adjusted to allow very few sessions in the zone. Other users that are connected with well-provisioned links will then be denied access, even if the bottleneck link would not be involved in those sessions. Other problems will occur when the usage behaviour is changed in some way, or when there are topology changes. Changed user behaviour could be that more users than usually gets their calls routed over a thin or loaded link, which could cause packet drops or increased delays. Topology change could be caused by link failure. This causes rerouting of packets meaning that the packets then take alternate paths through the network. Topology change can also be that a link characteristic is changed in some way, e.g. increased or decreased bandwidth, delay, etc.
Another problem is that gatekeeper-routed calls cannot be guaranteed high service quality in case direct calls are allowed. If a gatekeeper Gk performs bandwidth management for gatekeeper-routed calls in a zone and is unaware of simultaneous direct calls, the total traffic volume may exceed available bandwidth at some link. The problem here is that both gatekeeper-routed and direct calls use the same resources. This is due to that the gatekeeper Gk performs bandwidth management and approves bandwidth requests on gatekeeper-routed calls while direct calls can be made within the network without informing the gatekeeper about the bandwidth usage. In the case where direct calls are used within the IP telephony network, service differentiation, i.e. mechanisms in network elements that prioritise and forward important calls before less important calls, is necessary as soon as some sort of guarantees for a service is required. Gatekeeper approved calls can then be marked as important and forwarded first while direct calls are marked as less important.
For ongoing calls, problems might occur if there are additional endpoints that want to join the session and these endpoints are located on networks segments without available resources or where the available resources are not sufficient to provide predictable service. This issue will only occur when there is a multipart conference involving more than two endpoints.
Yet another problem is that different H.323 zones might be separated by non-H.323 enabled networks. Currently there are no means to provide a predictable service in this case because resources are not controlled in a non-H.323 network.
QoS support for H.323 using RSVP is currently under development. However, QoS support using RSVP is not scalable, especially not when there are calls made between endpoints in different zones where there are a non-H.323 enabled network in between. RSVP does per call signalling and reservations that would load the networks with signalling instead of useful traffic, i.e. media streams, and set-up per call state in routers.
Current H.323 systems do not allow reservations in advance, which makes it hard to plan meetings with predicted quality.
Yet another problem in the H.323 standard is that a bandwidth request is always approved or rejected. A more flexible approach between the bandwidth management functionality and the end-user is preferred.
The European patent document EP 0942560 discloses an apparatus and method for speech transport with adaptive packet size. It aims to minimize end-to-end delays caused by network traffic and low capacity routers in the network topology between two IP telephony devises. The aim is achieved by adopting packet sizes for speech transport. However, the document do not address how admission control can be done in speech transport systems, i.e. evaluating if there are sufficient capacity in the network before staring sessions and for communicating admission decisions to the system.